A voice communication technique by which superior echo canceling can be realized even in use of non-directional microphones is provided. When voice uttered by a near end speaker is defined as speaker's voice; sound obtained by emitting a received speech signal, which is a voice signal of a far end speaker, by a loudspeaker is defined as reproduction sound; a signal obtained by picking up an acoustic signal, which contains the speaker's voice and the reproduction sound, by a first microphone is defined as a first sound pickup signal; and a signal obtained by picking up an acoustic signal, which contains the speaker's voice and the reproduction sound, by a second microphone is defined as a second sound pickup signal, a voice communication device includes: a first subtraction unit that generates a first-subtracted first sound pickup signal that is a difference between a corrected sound pickup signal and the second sound pickup signal or the first sound pickup signal, the corrected sound pickup signal being obtained by correcting either one of the first sound pickup signal and the second sound pickup signal on the basis of an amplitude difference based on a difference between a distance d1 from the loudspeaker to the first microphone and a distance d2 from the loudspeaker to the second microphone (d2≤d1); and a second subtraction unit that generates a transmitted speech signal, which is to be transmitted to the far end speaker, by subtracting an adaptive-filtered second sound pickup signal, which is obtained by canceling echo, from the first-subtracted first sound pickup signal.

An echo cancelling system includes a data transmitter circuit and an echo canceller circuit. The data transmitter circuit is configured to receive a first transmitted signal. The first transmitted signal has a first sampling rate. The echo canceller circuit is configured to generate a second transmitted signal according to the first transmitted signal. The second transmitted signal has a second sampling rate. The second sampling rate is greater than the first sampling rate. The echo canceller circuit is further configured to generate an echo cancelling signal according to the second transmitted signal. The data transmitter circuit is further configured to generate an output signal according to a received signal and the echo cancelling signal.

A sound signal processing device, a sound signal processing method, and a sound signal processing program can remove echo signals included in input signals from a plurality of microphones with a simple circuit configuration are provided. The present device 1 includes an output part 20 that outputs a reception signal s2, an input part 30 that combines signals input from each of a plurality of microphones 3 and generates a transmission signal s4, wherein the microphones 3 collect an echo component of the reception signal and voice of a speaker and generate an echo signal es corresponding to the echo component and a sound signal s1 corresponding to the voice of the speaker, a removal signal generator 70 that generates a removal signal s5 removing the echo signal included in the transmission signal based on a filter coefficient F, a controller 50 that calculates the filter coefficient, and a removal part 80 that generates an echo removal signal s6 based on the transmission signal and the removal signal. The controller calculates individual filter coefficients k corresponding to the respective microphones, and calculate the filter coefficient by combining the individual filter coefficients.

A sound signal processing device, a sound signal processing method, and a sound signal processing program can remove echo signals included in input signals from a plurality of microphones with a simple circuit configuration are provided. The present device 1 includes an output part 20 that outputs a reception signal s2, an input part 30 that combines signals input from each of a plurality of microphones 3 and generates a transmission signal s4, wherein the microphones 3 collect an echo component of the reception signal and voice of a speaker and generate an echo signal es corresponding to the echo component and a sound signal s1 corresponding to the voice of the speaker, a removal signal generator 70 that generates a removal signal s5 removing the echo signal included in the transmission signal based on a filter coefficient F, a controller 50 that calculates the filter coefficient, and a removal part 80 that generates an echo removal signal s6 based on the transmission signal and the removal signal. The controller calculates individual filter coefficients k corresponding to the respective microphones, and calculate the filter coefficient by combining the individual filter coefficients.

A conferencing endpoint includes a loudspeaker component, a base microphone, and an adaptive volume control module which evaluates near-end audio conditions at a conferencing endpoint and automatically determines whether to change the audio volume at a near-end loudspeaker. The endpoint can, for example, detect that a person speaking into the base microphone has begun to speak more loudly to compensate for nearby background noise. The endpoint can automatically adjust the loudspeaker volume or provide an alert that the loudspeaker will be changed, giving the user the opportunity to accept or decline an increase in speaker volume.

There is provided a voice processing system capable of acquiring user voice more clearly, the voice processing system including: a mounting unit configured to be attached to a user. The mounting unit includes at least three voice acquisition units configured to acquire voice data for beamforming.

Techniques for announcing a communications session after the communications session is established between multiple user devices are described. In an example, a computer system may instruct a first user device to establish a communications session with a second user device. The computer system may receive, from the second user device, data indicating a request of the first user device for the communications session. Based at least in part on the data, the computer system may generate content associated with the first user device. The computer system may also instruct the second user device to accept the request and present the content after the communications session is established between the first user device and the second user device.

An apparatus for conducting acoustic echo cancellation by generating one or more error signals is provided. The apparatus is to generate a first echo estimate including one or more first echo estimation signals by filtering one or more reference signals using a first filter configuration, wherein the one or more reference signals correspond to one or more loudspeaker signals or are derived from the one or more loudspeaker signals. Moreover, the apparatus is to generate a second echo estimate including one or more second echo estimation signals by filtering the one or more first echo estimation signals using a second filter configuration; or, the apparatus is to generate the second echo estimate including the one or more second echo estimation signals by generating an intermediate signal from the one or more first estimation signals, and by filtering the intermediate signal using the second filter configuration. Furthermore, the apparatus is to generate the one or more error signals depending on one or more microphone signals and depending on the one or more second echo estimation signals. Moreover, the apparatus is to update the second filter configuration depending on the one or more error signals. Furthermore, the apparatus is to output the one or more error signals.

A first mobile communication device that includes a first microphone, a first speaker, and a first delay unit. The first microphone is configured to (i) receive, during a conference call, a first user first microphone signal from a first user, and (ii) output a first microphone digital signal to the first delay unit. The first user first microphone signal represents audio content outputted by the first user. The first delay unit is configured to delay, by a delay period, the first microphone digital signal to provide a delayed first user first device digital signal. The first mobile communication device is configured to output, to a mixer, the delayed first user first device digital signal. The delay period is determined based on measurements executed by at least one mobile communication device out of the first mobile communication device, a second mobile communication device and a third mobile communication device.

A method includes receiving a microphone audio signal and a playout audio signal, and determining a frequency representation of the microphone audio signal and a frequency representation of the playout audio signal. For each frequency representation, the method also includes determining features based on the frequency representation. Each feature corresponds to a pair of frequencies of the frequency representation and a period of time between the pair of frequencies. The method also includes determining that a match occurs between a first feature based on the frequency representation of the microphone audio signal and a second feature based on the frequency representation of the playout audio signal, and determining that a delay value between the first feature and the second feature corresponds to an echo within the microphone audio signal.